Method for providing VoIP services for wireless terminals

ABSTRACT

The present invention relates to a system and method for wireless telecommunication in a packet-based network comprising a Software Radio Port (SRP) which functions as a radio base station and a VoIP gateway to interconnect the wireless network with the VoIP packet network. Together with a Network Server Platform (NSP) and VoIP call-server, the SRP combines mobile call processing signaling with the VoIP call signaling to establish calls between the mobile and VoIP device or between mobiles. The SRP establishes the voice path to the mobile station over the air and the RTP media path to a party over a packet network for a call. These two paths are interconnected at the SRP so that an end-to-end voice path is established.

CROSS REFERENCE TO RELATED APPLICATIONS

This application is a continuation of U.S. patent application Ser. No.11/691,982 filed Mar. 27, 2007 now U.S. Pat. No. 7,664,103, which iscurrently allowed and is a continuation of U.S. patent application Ser.No. 10/010,682 filed Nov. 8, 2001, now U.S. Pat. No. 7,200,139, whereall of the above cited applications are incorporated herein by referencein their entirety.

FIELD OF THE INVENTION

The present invention relates to wireless telecommunications systemsand, more particularly, to a system and method of providing Voice overIP (VoIP) service in radio base stations for wireless terminals.

BACKGROUND OF THE INVENTION

Telephone calls have been traditionally routed through the publicswitched telephone network (PSTN) and, more recently, over data networksusing Voice over Internet Protocols (VoIP), the Internet being anexample of such a data network. VoIP transmission of voice conversationswith a VoIP-enabled phone serve to ease changes in the system, lowercosts and provide numerous new integrated services.

Wireless telephone calls are typically routed through end-to-end circuitswitching services using radio waves over the air and circuit-switchedlandline phone network. Wireless telecommunication systems contain aradio base station, which is a fixed device that enables communicationbetween the mobile transceiver and a landline phone network. There iscurrently no method or system for providing VoIP service via a radiobase station for a mobile terminal.

Therefore, there exists a need in the art to provide VoIP service via aradio base station for a wireless terminal so that a voice stream isonly circuit-switched over the air and packet-switched between the radiobase station and the VoIP phone network (i.e., a packet-based network).

SUMMARY OF THE INVENTION

In an exemplary embodiment of the present invention, a system and methodof wireless telecommunication in a packet-based network are providedcomprising receiving a call from a mobile station and converting thecall processing messages to VoIP protocol messages to set up a VoIP callto the destination. Once the called party answers, a two-way Real-TimeTransport Protocol (RTP) path will be established between the inventivesystem and the called party. A two-way voice path over the air is alsoestablished between the inventive system and the mobile station. Voicestreams from the mobile station will be converted to the RTP datapackets and transported to the called party. The RTP data packetsreceived from the called party will also be converted to the voiceframes to send to the mobile station over the air.

In another exemplary embodiment of the present invention, a system andmethod of wireless telecommunication in a packet-based network areprovided comprising receiving VoIP call processing messages from theVoIP phone network and converting the call processing messages to aspecific air interface protocol message to set up a call to the calledmobile station. Once the called mobile station answers, a two-way voicepath over the air is established between the inventive system and themobile station. Also a two-way Real-Time Transport Protocol (RTP) pathwill be established between the inventive system and the calling party.Then the inventive system will perform proper conversion for voicebetween the air interface and the packet-based interface.

The exemplary system and method comprise a Software Radio Port thatfunctions as a radio base station and a VoIP gateway, a VoIP call-Serverthat manages the call processing for VoIP calls, and a Network ServerPlatform that combines functions of a traditional Mobile SwitchingCenter (MSC) and a VoIP call-server control capability.

The exemplary system and method provide for VoIP service via a radiobase station for a wireless terminal so that a voice stream is onlycircuit-switched over the air and packet-switched over the land lines,with no need for circuit-switched land lines at the radio base station.

BRIEF DESCRIPTION OF THE DRAWINGS

A better understanding of the present invention will become apparentfrom the following detailed description of example embodiments and theclaims when read in connection with the accompanying drawings, allforming a part of the disclosure of this invention. While the foregoingand following written disclosure focus on disclosing example embodimentsof this invention, it should be clearly understood that the same is byway of illustration and example only and the invention is not limitedthereto. The spirit and scope of the present invention are limited onlyby the terms of the appended claims.

FIG. 1 illustrates an exemplary system architecture for the presentinvention.

FIG. 2 is a block diagram illustrating an exemplary Software Radio Port(SRP) of the present invention.

FIG. 3 is a block diagram illustrating an exemplary Network ServerPlatform (NSP) of the present invention.

FIG. 4 illustrates an exemplary call processing procedure involving acall from a mobile station (MS) to a VoIP (SIP) terminal.

FIG. 5 illustrates an exemplary call processing procedure involving acall from a VoIP (SIP) terminal to a mobile station (MS).

FIG. 6 illustrates an exemplary call processing procedure involving acall from a mobile station (MS1) to another mobile station (MS2).

FIG. 7 illustrates an exemplary call processing procedure involvingmobile station (MS)-initiated call release.

FIG. 8 illustrates an exemplary call processing procedure involvingnetwork-initiated call release.

FIG. 9 illustrates an exemplary call processing procedure involvingmobile handoff.

FIG. 10 illustrates an exemplary call processing procedure involving acall from a mobile station (MS) to a phone in the Public SwitchedTelephone Network (PSTN).

FIG. 11 illustrates an exemplary call processing procedure involving acall from a phone in the Public Switched Telephone Network (PSTN) to amobile station (MS).

DETAILED DESCRIPTION OF THE INVENTION

Before beginning a detailed description of the invention, it should benoted that, when appropriate, like reference numerals and characters maybe used to designate identical, corresponding or similar components indiffering figure drawings. Further, in the detailed description tofollow, example embodiments and values may be given, although thepresent invention is not limited thereto.

The present invention provides a method and system that provide VoIPservice for a wireless terminal. The VoIP service is provided via aradio base station wherein a received voice stream is circuit-switchedover the air. The voice stream received from a mobile station (MS) isconverted to RTP data packets at the radio base station and transportedover a packet-switched network to a desired destination. Conversely, theradio base station converts received RTP data packets to the propervoice stream for delivery over the air to a mobile station. Differentair interfaces (radio interfaces), such as GSM, IS136, CDMA, etc, can besupported by the radio base station. Different VoIP protocols can alsobe supported at the radio base station, such as H.323 and SIP.

FIG. 1 is a block diagram illustrating an exemplary embodiment of thepresent invention. A wireless terminal or mobile station (MS) 20 may bean IS-136 Terminal 17, a GSM terminal 16, etc. It is to be understoodthat the present invention is not so limited as a variety ofcapabilities may be used such as, but not limited to, GPRS-136 HS/EDGE,802.11 wireless LAN, etc.

The wireless terminal or mobile station may communicate over the airwith a Software Radio Port (SRP) 15. The SRP 15 combines functions of atraditional radio base station and a VoIP gateway (media and signalinggateway) to manage the air interface and packet network interface,respectively. These two interfaces are interconnected at the SRP 15.

A Network Server Platform (NSP) 12 may combine functions of atraditional Mobile Switching Center (MSC) and a VoIP call-server controlcapability.

The exemplary system illustrated in FIG. 1 may also contain a VoIPCall-Server 11. The VoIP Call-Server 11 is the call server for the VoIPnetwork and may handle requests or messages from a VoIP client. The VoIPclient may be, for example, a VoIP-enabled phone. The VoIP Call-Server11 may also manage requests or messages from a VoIP Gateway. The VoIPCall-Server 11 works with the NSP to obtain proper routing informationand forward received messages to desired destinations with/withoutmodification on the messages.

A PSTN/VoIP Gateway 23 may also be used to interconnect the PSTN withthe VoIP network. The Gateway 23 performs proper signaling and mediaconversion. Finally, a VoIP enabled phone, such as a SIP enabled phone,may be used. Although the exemplary embodiment demonstrates SIP tosupport the VoIP, the present invention is not so limited as anysuitable VoIP protocol may be used such as H.323 to implement the VoIPservice, for example.

FIG. 2 is a functional block diagram of an exemplary SRP 15. The airinterface 200 may be an interface to support the IS-136 air interfacefor voice services. The SRP 15 may be further equipped with VoIPcapability with a VoIP Media Gateway 205 and VoIP Signaling Gateway 207,and with an IP/Ethernet interface 210 that integrates the SRP 15 intothe core packet data network for packet data service.

The VoIP Signaling Gateway may manage the VoIP call processing. Itinterconnects the mobile call control component with the VoIP callcontrol. The VoIP Media Gateway 205 may perform voice codec translation.In an uplink direction, the input voice stream may be received at theair interface 200 and then forwarded to the VoIP Media Gateway 205. Inthe VoIP Media Gateway 205, the data is packetized and transported viathe IP/Ethernet interface 210 to the desired destination. In thedownlink direction, voice data packets may be received via theIP/Ethernet interface 210 at the VoIP Media Gateway 205. The datapackets are converted and sent through the air interface 200 to themobile station, such as a GSM terminal 16 or IS-136 Terminal 17.

The SRP 15 may further comprise a call control & OAM (Operation,Administration and Maintenance) 206. The call control component controlsmobile call processing such as IS-136 call processing and coordinateswith the VoIP signaling Gateway 207 for VoIP call related operations,such as requesting the VoIP signaling gateway 207 to set up a VoIP callwhile an IS-136 mobile is making a call. It also coordinates with theVoIP Media Gateway 205 such as in instructing the VoIP Media Gateway toset up an RTP media path at the proper call setup stage. The OAMcomponent provides necessary functions to ensure the normal operation ofthe SRP 15.

FIG. 3 is a functional block diagram illustrating an exemplary NetworkServer Platform (NSP) 12, which may provide SRP control, callmanagement, and mobility management. The NSP 12 may comprise a MobileSwitching Center/Visitors' Location Register (MSC/VLR) 305. The MSC/VLR305 may perform call control-related operations for IS136 mobilesincluding but not limited to Mobile Station registration, callorigination, call termination and handoff. The MSC/VLR 305 may managethe SRP 15. The MSC/VLR 305 may also manage the VoIP Call-Server 11 viathe VoIP Call-Server Control 308.

The NSP 12 may further comprise a Home Location Register (HLR) 315,which is a mobile subscriber database and authentication center forcaller verification. The HLR 315 may identify and/or verify a subscriberand may also contain a subscriber database related to features andservices. The HLR also has the current mobile location information foreach of the mobiles. The NSP 12 also contains a VoIP Call-Server Control308, which controls VoIP call processing related operations incoordination with Mobile Switching Center (MSC) call control.

The NSP 12 may also comprise an IP interface 301 which may be anIP/Ethernet Interface that connects the NSP to the packet network. TheIP interface 301 is responsible for MSC/VLR 305 to SRP 15 and VoIPCall-Server Control 308-to-VoIP Call-Server 11 communication.

The system and method of the present invention may be better understoodthrough the following described exemplary embodiments. It should benoted that although the invention is described with reference toillustrative embodiments thereof, the present invention is not solimited and that numerous other modifications and embodiments may bedevised.

In one exemplary embodiment as illustrated in FIG. 4, a two-way RTPmedia path may be set up via an RTP pair of ports, i.e., one for RTP andthe other for Real-Time Control Protocol (RTCP), between a VoIP Phone,such as the SIP phone 10, and the SRP 15. The RTP is for voice packetsand RTCP is used to transmit control packets to participants from timeto time regarding a particular RTP session. A two-way voice path is setup over the air between the SRP 15 and the Mobile Station (MS) 20. Thesetwo paths are interconnected at the SRP 15. In this example, a VoIPphone number is dialed at the MS 20, for example a wireless device suchas a GSM Terminal 16 or IS-136 Terminal 17. The VoIP phone may be of anyone of many VoIP phone, for example, Session Initiation Protocol (SIP)phone. Upon the dialing of the VoIP phone number, an IS-136 CallOrigination Message, for example, may be sent to the SRP 15 which mayforward the Call Origination message to the NSP 12 (step 401). The NSP12 may respond with a Call Setup message to the SRP 15 (step 402) whichmay then send an IS-136 Digital Traffic Channel (DTC) Designationmessage to tune the Mobile Station (MS) 20 to the DTC (step 403). If theSRP 15 is unable to detect that the MS 20 is on the DTC, the call may beabandoned before any SIP-related message exchange occurs. After the SRP15 detects the MS 20 as being on the DTC (step 404), the SRP 15 sends aSIP INVITE message to the SIP server 11 (step 405). If there is noresponse to the INVITE message from the SRP 15, the SRP 15 may time outand re-send the message.

The SIP server 11 may announce the incoming call to the NSP 12 (step406), which may analyze the called number and realize it is not a mobilesubscriber. The NSP 12 may respond back to the SIP server (step 407)which may then analyze the number and realize it is a local phone andforward the INVITE message to the called VoIP phone such as the SIPphone 10. The SIP phone 10 may optionally respond to the INVITE messagewith a SIP 100 TRYING message. The SIP phone may then send a 180 RINGINGmessage to the SIP Server 11 which then may relay the message to the SRP15 (step 408). The SRP 15 may then generate a ring back tone, which maybe heard at the MS 20 (step 409).

When the SIP phone 10 is answered, an SIP 200 OK message may be sent tothe SIP server 11 which then may forward the message to the SRP 15 (step410). The SRP 15 may respond with an ACK message and may then set up theRTP ports for sending and receiving the RTP/RTCP packets (step 411).

Thus, a two-way RTP media path is set up between the SIP Phone 10 andthe SRP 15. The SRP 15 then interconnects the RTP path with the two-waydigital traffic channel between the SRP 15 and the MS 20 so that anend-to-end voice path between the MS 20 and SIP phone 10 has beenestablished. The SRP 15 may exchange optional send/receive Real TimeControl Protocol (RTCP) reports with the SIP phone 10 (step 413) and maysend Call Connect messages to the NSP 12 (step 414). If the SIP phone 10is busy, a SIP 4xx Failure message may be sent to the SRP 15 which maydisconnect the call. If the SIP phone 10 does not answer, the SRP 15 maytime out on receiving the OK message and may send a SIP 4xx Failuremessage to the SIP Server 11 and disconnect the call.

In another exemplary embodiment as illustrated in FIG. 5, a two-wayvoice path is established between a VoIP phone, such as a SIP phone 10,and a mobile station (MS) 20 via a SRP 15, wherein the call originatesat the VoIP phone (i.e., SIP phone) and terminates at the MS 20. In thisexample, a VoIP phone, for example a SIP phone 10, may place a call to amobile station (MS) 20 by sending an INVITE message to a VoIPcall-server (illustrated in FIG. 5 as a SIP server 11) (step 501). TheSIP server 11 may send a message containing the called number to the NSP12 (step 502). The NSP 12 may then analyze the called number and realizethat the called number is a registered mobile station covered by the SRP15 and may send a page request message to the SRP 15. The SRP 15 maythen forward the page request message to the MS 20 (step 503) and mayalso send back a page response received from the MS 20 to the NSP 12(step 504). The page request may be an IS-136 message but is not solimited. If the MS 20 does not respond to the page request message, theSRP 15 may time out and notify the SIP Server 11 through the NSP 12. Inthis case, the SIP Server 11 may abandon the call and may send a SIP 4xxFailure message to the calling SIP phone 10. However, if the MS 20properly responds to the page request message, the NSP 12 may theninstruct the SIP server 11 to send the INVITE message to the SRP 15(step 505), which may optionally respond with a SIP 100 TRYING messageto the SIP phone 10 (step 506). The SRP 15 further sends Digital TrafficChannel (DTC) Designation messages to the MS 20, such as but not limitedto IS-136 messages, and may detect when the MS 20 is tuned to the DTC(step 507). If the SRP 15 cannot detect that the MS 20 is on the DTC,the call may be abandoned and the SRP 15 may send a SIP 4xx Failuremessage to the SIP server 11. However, when the MS 20 is on the DTC andis detected as being on the DTC, the SRP 15 may send an alert message(such as an IS-136 alert) to the MS 20 (step 508). The SRP 15 receivesan ACK message from the MS 20 and sends a SIP 180 RINGING message to theSIP Server 11. The SIP server 11 may then forward the 180 RINGINGmessage to the SIP phone 10 (step 509). When the MS 20 answers the call,the SRP 15 may then send an SIP 200 OK message to the SIP server 11,which forwards the SIP 200 OK message to the SIP phone 10 (step 510).The SIP phone 10 may then send an ACK message to the SIP Server 11 (step511) and a two-way voice path is established. The two parties, i.e. theSIP phone 10 and the SRP 15 may then exchange optional Real Time ControlProtocol (RTCP) send/receive reports (step 513) and the SRP 15 may senda CONNECT message to the NSP 12 (step 514).

In the exemplary embodiment illustrated in FIG. 6, the exemplaryembodiment of FIG. 4 and FIG. 5 are combined such that both the callingparty and the called party are mobile stations (MS1 20 and MS2 21,respectively). The calling MS1 20 and called MS2 21 may be registeredeither on the same or different SRPs, such as SRP1 14 and SRP2 15. Inthis example, one mobile station (MS1) 20 initiates a call by sending acall origination message to a first SRP (SRP1) 14. SRP1 14 processes andtransmits the call origination message to the NSP 12, which responds byreturning a call setup message back to the SRP1 14 (step 602). SRP1 14further sends Digital Traffic Channel (DTC) Designation messages to theMS1 20, such as but not limited to IS-136 messages, and may detect whenthe MS1 20 is tuned to the DTC (step 604). If the SRP1 14 cannot detectthat the MS1 20 is on the DTC, the call may be abandoned. Otherwise,when the MS1 20 is on the DTC and is detected as being on the DTC, SRP114 may send an INVITE message to the SIP Server (step 605). The SIPserver 11 may announce the incoming call to the NSP 12 (step 606). TheNSP 12 may then analyze the called number and realize that the callednumber is a registered mobile station covered by a second SRP (SRP2) 15and may send a page request message to the SRP2 15. SRP2 15 may thenforward the page request message to the second mobile station (MS2) 21(step 606) and may also send back a page response received from the MS221 to the NSP 12 (step 607). The page request may be an IS-136 messagebut is not so limited. If the MS2 21 does not respond to the pagerequest message, the SRP2 15 may time out and notify the SIP Server 11through the NSP 12. In this case, the SIP Server 11 may abandon the calland may send a SIP 4xx Failure message to the calling SRP1 14. However,if the MS2 21 properly responds to the page request message, the NSP 12may then instruct the SIP server 11 to send the INVITE message to theSRP2 15 (step 608), which may optionally respond with a SIP 100 TRYINGmessage to the SIP server 11 (step 609). The SRP2 15 further sendsDigital Traffic Channel (DTC) Designation messages to the MS2 21, suchas but not limited to IS-136 messages, and may detect when the MS2 21 istuned to the DTC (step 610). If the SRP2 15 cannot detect that the MS221 is on the DTC, the call may be abandoned and the SRP2 15 may send aSIP 4xx Failure message to the SIP server 11 (not shown). However, whenthe MS2 21 is on the DTC and is detected as being on the DTC, the SRP215 may send an alert message (such as an IS-136 alert) to the MS2 21(step 611). The SRP2 15 receives an ACK message from the MS2 21 andsends a SIP 180 RINGING message to the SIP Server 11. The SIP server 11may then forward the SIP 180 RINGING message to the SRP1 14 (step 612),which generates and sends a ring back tone to the MS1 20. When MS2 21 isanswered, the SRP2 15 sends an SIP 200 OK message to the SIP server 11,which then may forward the message to the SRP1 14 (step 613). The SRP114 may respond with an ACK message and may then set up the RTP ports forsending and receiving the RTP packets (step 614). Thus, the two-wayvoice path is set up between the MS1 20 and the MS2 21 via SRP1 14 andSRP2 15 (step 615). This two-way voice path consists of: two-way digitaltraffic channel between MS1 20 and SRP 1 14, two-way RTP path betweenthe SRP1 14 and SRP2 15, and two-way digital traffic channel between MS221 and SRP2 15. The SRP1 14 may exchange optional send/receive Real TimeControl Protocol (RTCP) reports with the SRP2 15 (step 616) and both maysend Call Connect messages to the NSP 12, that is, SRP1 14 may send CallConnect messages to the NSP 12 (step 617) and SRP2 15 may send CallConnect messages to the NSP 12 (step 618).

In another exemplary embodiment illustrated in FIG. 7, a call isdisconnected by the mobile station (MS) 20. In this example, the mobilestation (MS) 20 hangs up and an IS-136 release message, for example, maybe sent to the SRP 15 which responds with a IS-136 Base Station ACKmessage and releases the radio resource and the RTP media path (step701). The SRP 15 also may send a SIP BYE message to a VoIP call-server,in this case, the SIP server 11, which may pass the BYE message to aVoIP phone, such as the SIP phone 10 (step 702). The SIP phone 10 mayrespond by ending the call and returning a SIP 200 OK message to the SRP15 (step 702) via the SIP Server 11. After receiving the 200 OK message,the SRP 15 sends a Call Release message to the NSP 12, which may thenupdate the call state of the MS 20 (step 703). Alternatively, if the SIP200 OK message is not received at the SRP 15 after sending the BYEmessage, the SRP 15 will time out after a predetermined period of timeand may send a call Release message to the NSP 12 after thepredetermined period of time has elapsed. The NSP 12 may then send anacknowledge message back to the SRP 15 (step 704). If the acknowledgemessage is not received at the SRP 15, the SRP 15 may try again for apredetermined number of times.

In another exemplary embodiment as illustrated in FIG. 8, calldisconnection is initiated in the network. Such may occur, for example,when the party from the network side decides to disconnect the call orwhen the SIP server decides to release the call due to error or othervarious reasons. In this example, the VoIP call-server or SIP Server 11receives a BYE message from the SIP Phone 10 and forwards the message tothe SRP 15 (step 801). The SRP 15 receives the BYE message and respondswith a SIP 200 OK message to the SIP server 11, disconnects the RTPmedia path, and sends an MS release message (e.g., an IS-136 releasemessage) to the mobile station (MS) 20 (step 802). After receiving theACK message from the mobile station, the SRP 15 sends a Call Releasemessage to the NSP 12 (step 803) which returns an acknowledge message(step 804). If the SRP 15 does not receive the acknowledge message fromthe MS 20, the SRP 15 may continue to re-try a predetermined number oftimes. Also, if the SRP 15 does not receive the acknowledge message fromthe NSP 12, the SRP 15 may re-try a predetermined number of times.

In another exemplary embodiment of the present invention as illustratedin FIG. 9, a call is handed off from an old SRP 14 to a new SRP 15. Inthis example, the mobile station (MS) 20, while engaged with a partywithin range of the old SRP 14, moves out of range of the old SRP 14 andinto range of the new SRP 15. Channel quality messages being sent to theold SRP 14 from the MS 20 may indicate that the error rate has reachedthe handoff point (step 901)—i.e., because the MS 20 is moving out ofrange of the old SRP 14, the channel quality is attenuating to the pointthat handoff should occur to maintain the connection. At this point, theold SRP 14 may then send a Handoff Request message along with a list ofhandoff candidates (the Mobile Assisted Hand-Off list—MAHO list, forexample) to the NSP 12 (step 902). Using MAHO as an example, the MS 20may assist in assigning a voice channel by reporting its surroundingbase stations' signal strengths to the current base station, forexample. The NSP 12 may then verify the availability of a resource fromthe MAHO list, for example, and may send a Handoff Preparation messageto a new SRP 15 (step 903). The NSP 12 may abort the handoff if noavailable resources are identified. If resources are available, however,the new SRP 15 may activate a new traffic channel and then send amessage to the old SRP 14 to handoff the mobile station (MS) 20 to thenew traffic channel (step 904). After receiving the message from the newSRP 15, the old SRP 14 sends a Handoff message to the MS 20 (step 905)(i.e., an IS-136 Dedicated DTC handoff message), which responds back anACK to the old SRP 14. If the old SRP 14 does not receive this ACKresponse, however, the old SRP 14 may send a handoff failure message tothe NSP 12 and the NSP 12 may inform the new SRP 15 to release allocatedresources. However, if the response is received at the old SRP 14, theold SRP 14 may then instruct the NSP 12 to transfer the call to the newSRP 15 and may also deactivate the traffic channel and release otherresources (step 906). The NSP may then send a message to the new SRP 15for a conference VoIP call and also may send a message to the old SRP 14to release the VoIP call (step 907). When the new SRP 15 detects thatthe MS 20 is on the new traffic channel, it may then send a SIP INVITEmessage to the SIP server 11 (step 908) which may send a message to NSP12 for called number analysis (step 909). The NSP 12 analyzes the callednumber and responds back to the SIP server 11 after it realizes that thecalled number is not a mobile station (step 910). The SIP server 11further finds out if the called number is a local SIP phone and forwardsthe INVITE message to the SIP phone 10 (step 911) which sends back a 180RINGING message. The SIP server 11 relays the 180 RINGING message to SRP15 (step 912). The new SRP 15 does not generate a ring back tone. TheSIP phone 10 may send a SIP OK message to the SIP server 10 whichforwards it to the SRP 15 (step 913). The SRP 15 may respond with a SIP200 ACK message back to the SIP Server 11 which forwards it to the SIPphone 10 (step 914). A voice path between the MS 20 and the SIP Phone 10is established and is interconnected by the new SRP 15 (step 915). Thenew SRP 15 may send a message to the NSP 12 to indicate that the handoffis complete (step 916). The old SRP 14 receives the Release old Callmessage from the NSP 12 and may then send a BYE message to the SIP phone10 via the SIP server 11 (step 917). The SIP phone 10 returns a SIP 200OK message (step 918) and the old SRP 14 sends a CALL Release message toinform the NSP 12 that the call has been released by the old SRP 14(step 919). The NSP 12 acknowledges with an ACK message (step 920).

In another exemplary embodiment of the present invention illustrated inFIG. 10, a two-way voice path may be set up between a telephony networksuch as a Public Switched Telephone Network, such as a PSTN phone 24,and the Mobile Station (MS) 20 via the SRP 15 with the call originatingfrom the Mobile Station (MS) 20. It is understood that although theillustrative embodiment demonstrates a voice path between a phone in thePSTN network and a mobile station, the present invention is not solimited and may be used with any telephony network including PrivateBranch Exchange system (PBX), for example. In this example, a phonenumber is dialed at the MS 20, for example a wireless device such as aGSM Terminal 16 or IS-136 Terminal 17. Upon the dialing of the phonenumber, a Call Origination Message, for example an IS-136 callorigination message, may be sent to the SRP 15 which may forward theCall Origination message to the NSP 12 (step 1001). The NSP 12 mayrespond with a Call Setup message to the SRP 15 (step 1002) which maythen send a Digital Traffic Channel (DTC) Designation message (e.g., anIS-136 Digital Traffic Channel) to effect tuning of the Mobile Station(MS) 20 to the DTC (step 1003). After the SRP 15 detects the MS 20 asbeing on the DTC (step 1004), the SRP 15 sends a SIP INVITE message tothe SIP server 11 (step 1005). However, if the SRP 15 is unable todetect that the MS 20 is on the DTC, the call may be abandoned beforeany SIP-related message exchange occurs. Also, if there is no responseto the INVITE message from the SRP 15, the SRP 15 may time out andre-send the message several times.

The SIP server 11 may announce the incoming call to the NSP 12 (step1006), which may analyze the called number. The NSP 12 realizes thecalled number is not a subscriber and responds back to the SIP server.The SIP server 11 further analyzes the called number and realizes it isnot a local phone number. The SIP server 11 then forwards the INVITEmessage to a SIP Gateway 23 where all necessary SIP/PSTN interworkingfunctions are performed (step 1007). Call processing occurs in the PSTNafter an IAM (Initial Address Message) message is received at the PSTNfrom the SIP Gateway. The SIP Gateway 23 may further respond to theINVITE message by sending a SIP 100 TRYING message to the SIP Server 11(step 1008) or may receive an ACM (Address Complete Message) message(step 1009) from the PSTN phone 24 and send a SESSION PROGRESS messageto the SIP Server 11 which may relay these messages to the SRP 15. Alsoa one-way voice path is established from the PSTN phone 24 to the MS 20via the SIP Gateway 23 and SRP 15 so that a ring back tone may be heardat the MS 20.

When the PSTN phone 24 is answered, an ANM (Answer Message) message maybe sent to the SIP Gateway 23 (step 1010) which then may send an OKmessage to the SIP Server 11 which may relay the message to the SRP 15.The SRP 15 may respond with an ACK message and may then set up the RTPports for sending and receiving the RTP packets (step 1011).

Thus, the two-way voice path is set up between the PSTN phone 24 and theMS 20 via the SIP Gateway 23 and SRP 15 (step 1012). The SRP 15 mayexchange optional send/receive Real Time Control Protocol (RTCP) reportswith the SIP Gateway 23 (step 1013) and may send Call Connect messagesto the NSP 12 (step 1014). If the PSTN phone 24 is busy, a SIP 4xxFailure message may be sent to the SRP 15, which may disconnect thecall. If the PSTN phone 24 does not answer, the SRP 15 may time out onreceiving the 200 OK message and may send a SIP 4xx Failure message tothe SIP Server 11 and disconnect the call.

In another exemplary embodiment as illustrated in FIG. 11, a two-wayvoice path is established between a PSTN phone 24 and a mobile station(MS) 20 wherein the call originates at the PSTN phone 24 and terminatesat the MS 20. It is understood that although the illustrative embodimentdemonstrates a voice path between a phone in the PSTN and a mobilestation, the present invention is not so limited and may be used withany telephony network including Private Branch Exchange system (PBX),for example. In this example, a call is placed to a mobile station (MS)20 via a VoIP Gateway (such as but not limited to a SIP Gateway 23 asillustrated in FIG. 11) by sending an IAM message from the PSTN phone 24to the SIP Gateway 23 which then sends an INVITE message to a VoIPcall-server (illustrated in FIG. 11 as SIP server 11) (step 1101). TheSIP server 11 may send a message containing the called number to the NSP12 (step 1102). The NSP 12 may then analyze the called number andrealize that the called number is a registered mobile station covered bythe SRP 15 and may send a page request message to the SRP 15. The SRP 15may then forward the page request message to the MS 20 (step 1103) andmay also send back a page response received from the MS 20 to the NSP 12(step 1104). The page request may be an IS-136 message but is not solimited. If the MS 20 does not respond to the page request message, theSRP 15 may time out and notify the SIP Server 11 through the NSP 12. Inthis case, the SIP Server 11 may abandon the call and may send a SIP 4xxFailure message to the calling SIP Gateway 23. Otherwise, if the MS 20properly responds to the page request message, the NSP 12 may theninstruct the SIP server 11 to send the INVITE message to the SRP 15(step 1105), which may optionally respond with a SIP 100 TRYING messageto the SIP Server 11 which then forwards the message to the SIP Gateway23 (step 1106). The SRP 15 further sends Digital Traffic Channel (DTC)Designation messages to the MS 20, such as but not limited to IS-136messages, and may detect when the MS 20 is tuned to the DTC (step 1107).If the SRP 15 cannot detect that the MS 20 is on the DTC, the call maybe abandoned and the SRP 15 may send a SIP 4xx Failure message to theSIP server 11 (not shown). Otherwise, when the MS 20 is on the DTC andis detected as being on the DTC, the SRP 15 may send an alert message(such as an IS-136 alert) to the MS 20 (step 1108). The SRP 15 receivesan ACK message from the MS 20 and sends a SIP 180 RINGING message to theSIP Server 11. The SIP server 11 may then forward the 180 RINGINGmessage to the SIP Gateway 23 (step 1109), which sends an ACM message tothe PSTN phone 24. When the MS 20 answers the call, the SRP 15 may thensend an OK message to the SIP server 11 (step 1110) which forwards theOK message to the SIP Gateway 23. The SIP Gateway 23 then sends an ANMmessage to the PSTN phone 24. The SIP Gateway 23 sends an ACK message tothe SIP Server 11 which then forwards the ACK message to the SRP 15(step 1111) and a two-way voice path is established. The SIP Gateway 23and the SRP 15 may then exchange optional Real Time Control Protocol(RTCP) send/receive reports (step 1113) and the SRP 15 may send aCONNECT message to the NSP 12 (step 1114).

This concludes the description of the example embodiments. Although thepresent invention has been described with reference to illustrativeembodiments thereof, it should be understood that numerous othermodifications and embodiments can be devised by those skilled in the artthat will fall within the scope and spirit of the principles of theinvention. More particularly, reasonable variations and modificationsare possible in the component parts and/or arrangements of the subjectcombination arrangement within the scope of the foregoing disclosure,the drawings and the appended claims without departure from the spiritof the invention. In addition to variations and modifications in thecomponent parts and/or arrangements, alternative uses will also beapparent to those skilled in the art.

1. A device for processing voice over internet protocol data packets for wireless terminals, comprising: an internet protocol/ethernet interface for establishing a real-time transport protocol path with a voice over internet protocol phone; a voice over internet protocol media gateway interposed between an air interface that establishes a voice path wirelessly with a mobile station and the internet protocol/ethernet interface for media conversion and transportation; a voice over internet protocol signaling gateway for controlling voice over internet protocol call processing; and, a call control in communication with the voice over internet protocol media gateway and the voice over internet protocol signaling gateway for controlling call processing for wireless terminals and coordinating with voice over internet protocol call processing.
 2. The device of claim 1, wherein signaling messages and a voice stream are received from the mobile station via the air interface.
 3. The device of claim 1, wherein the call control is configured for: receiving signaling messages via the air interface; instructing the voice over internet protocol media gateway to set up real-time transport protocol paths to called parties; and, instructing the voice over internet protocol signaling gateway to set up voice over internet protocol calls to the called parties.
 4. The device of claim 1, wherein the voice over internet protocol signaling gateway is configured for: receiving messages from the call control; processing the messages from the call control; and, managing voice over internet protocol call-related activities.
 5. The device of claim 1, wherein the voice over internet protocol media gateway is configured for: receiving messages from the call control; processing the messages from the call control; receiving a voice stream via the air interface; and packetizing the voice stream into real-time transport protocol data packets.
 6. The device of claim 1, wherein the internet protocol/ethernet interface receives real-time transport protocol data packets from the voice over internet protocol media gateway and messages from the call control and the voice over internet protocol signaling gateway, and sends the real-time transport protocol data packets and the messages to a packet data network.
 7. The device of claim 1, wherein the internet protocol/ethernet interface receives messages and real-time transport protocol packets from a packet data network, sends the real-time transport protocol packets to the voice over internet protocol media gateway, and sends the messages to the call control and the voice over internet protocol signaling gateway.
 8. The device of claim 1, wherein the call control is configured for: receiving signaling messages from the internet protocol/ethernet interface and the voice over internet protocol signaling gateway; and managing mobile station-related activities.
 9. The device of claim 1, wherein the voice over internet protocol signaling gateway is configured for: receiving messages from the internet protocol/ethernet interface; instructing the call control to manage mobile calls; and managing voice over internet protocol call-related activities.
 10. The device of claim 1, wherein the voice over internet protocol media gateway is configured for: receiving the voice over internet protocol data packets from the internet protocol/ethernet interface; and converting the voice over internet protocol data packets to a voice stream.
 11. The device of claim 1, wherein the air Interface receives a voice stream from the voice over internet protocol media gateway and receives signaling messages from the call control.
 12. A method of providing a path between a voice over internet protocol device in a network and a mobile station, comprising: tuning the mobile station to a digital traffic channel to establish a voice path wirelessly via a software radio port; engaging a voice over internet protocol call-server to set up a voice over internet protocol call; generating a ringback tone to the mobile station; establishing a real-time transport protocol media path for exchange of real-time transport protocol data packets via said the software radio port; and interconnecting the voice path and the real-time transport protocol media path over a packet network via the software radio port.
 13. The method of claim 12, wherein the voice over internet protocol device comprises a voice over internet protocol phone.
 14. The method of claim 12, wherein the tuning is in response to receiving a call origination message from the mobile station and engaging a network server platform to set up a mobile call.
 15. The method of claim 12, wherein the tuning comprises: sending a message to tune the mobile station to a specified digital traffic channel; and detecting the mobile station as being tuned to the specified digital traffic channel.
 16. The method of claim 12, wherein the engaging comprises: sending a voice over internet protocol call connection request to the voice over internet protocol call-server; analyzing a called number; and setting up a voice over internet protocol call via the voice over internet protocol call-server.
 17. The method of claim 12, wherein the generating comprises: receiving a ringing indication from a called party; generating the ringback tone in response to the receiving; and transmitting the ringback tone to the mobile station.
 18. The method of claim 12, wherein the establishing comprises: receiving a connect indication from a called party; turning off the ringback tone; setting up the real-time transport protocol media path for exchange of the real-time transport protocol data packets; and informing a network server platform of the call connection.
 19. The method of claim 12, wherein the interconnecting comprises: converting received voice frames to real-time transport protocol packets to be sent to the packet network, and converting received real-time transport protocol packets to voice frames to be sent to the mobile station.
 20. A method of providing a path between a voice over internet protocol device in a packet network and a mobile station, comprising: processing a call connection request at a voice over internet protocol call-server; initiating mobile call set-up at a network server platform; tuning the mobile station to a digital traffic channel to establish a first path wirelessly via a software radio port; establishing a real-time transport protocol media path for exchange of real-time transport protocol data packets via the software radio port; and interconnecting the first path and the real-time transport protocol media path over the packet network via the software radio port. 